December 2008 - Freeswitch-svn - FreeSWITCH Mailing Lists

[commit] r11000 - freeswitch/branches/gmaruzz/src/mod/endpoints/mod_skypiax
by FreeSWITCH SVN 30 Dec '08

30 Dec '08

Author: gmaruzz Date: Tue Dec 30 06:53:18 2008 New Revision: 11000 Log: skypiax: now makes outgoing calls and close them almost correctly Modified: freeswitch/branches/gmaruzz/src/mod/endpoints/mod_skypiax/mod_skypiax.c Modified: freeswitch/branches/gmaruzz/src/mod/endpoints/mod_skypiax/mod_skypiax.c ============================================================================== --- freeswitch/branches/gmaruzz/src/mod/endpoints/mod_skypiax/mod_skypiax.c (original) +++ freeswitch/branches/gmaruzz/src/mod/endpoints/mod_skypiax/mod_skypiax.c Tue Dec 30 06:53:18 2008 @@ -511,7 +511,8 @@ switch_core_session_add_stream(*new_session, NULL); if ((tech_pvt = (private_t *) switch_core_session_alloc(*new_session, sizeof(private_t))) != 0) { - tech_pvt->p = &globals.SKYPIAX_INTERFACES[2]; //FIXME + tech_pvt->p = &globals.SKYPIAX_INTERFACES[19]; //FIXME + globals.SKYPIAX_INTERFACES[19].tech_pvt = tech_pvt; //FIXME channel = switch_core_session_get_channel(*new_session); skypiax_tech_init(tech_pvt, *new_session, NULL); } else { @@ -588,8 +589,17 @@ if (channel) { switch_channel_hangup(channel, SWITCH_CAUSE_NORMAL_CLEARING); //DEBUGA_SKYPE("after channel_on_hangup\n", SKYPIAX_P_LOG); - } - } + + } else { + ERRORA("no channel?\n", SKYPIAX_P_LOG); + } + + } else { + ERRORA("no session?\n", SKYPIAX_P_LOG); + } + + } else { + ERRORA("no tech_pvt?\n", SKYPIAX_P_LOG); } p->interface_state = SKYPIAX_STATE_DOWN; global_session = NULL; //FIXME

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[commit] r10999 - freeswitch/trunk/libs/esl
by FreeSWITCH SVN 30 Dec '08

30 Dec '08

Author: mikej Date: Mon Dec 29 20:55:59 2008 New Revision: 10999 Log: set default level Modified: freeswitch/trunk/libs/esl/fs_cli.c Modified: freeswitch/trunk/libs/esl/fs_cli.c ============================================================================== --- freeswitch/trunk/libs/esl/fs_cli.c (original) +++ freeswitch/trunk/libs/esl/fs_cli.c Mon Dec 29 20:55:59 2008 @@ -274,7 +274,7 @@ signal(SIGINT, handle_SIGINT); handle.debug = 0; - esl_global_set_default_logger(2); /* default debug level to 2 */ + esl_global_set_default_logger(3); /* default debug level to 3 (error) */ for(;;) { int option_index = 0;

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[commit] r10998 - freeswitch/trunk/libs/esl
by FreeSWITCH SVN 30 Dec '08

30 Dec '08

Author: silik0n Date: Mon Dec 29 20:24:32 2008 New Revision: 10998 Log: tweak Modified: freeswitch/trunk/libs/esl/fs_cli.c Modified: freeswitch/trunk/libs/esl/fs_cli.c ============================================================================== --- freeswitch/trunk/libs/esl/fs_cli.c (original) +++ freeswitch/trunk/libs/esl/fs_cli.c Mon Dec 29 20:24:32 2008 @@ -274,7 +274,7 @@ signal(SIGINT, handle_SIGINT); handle.debug = 0; - esl_global_set_default_logger(0); /* default debug level to 0 */ + esl_global_set_default_logger(2); /* default debug level to 2 */ for(;;) { int option_index = 0;

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[commit] r10997 - freeswitch/trunk/libs/esl
by FreeSWITCH SVN 30 Dec '08

30 Dec '08

Author: anthm Date: Mon Dec 29 19:31:15 2008 New Revision: 10997 Log: update Modified: freeswitch/trunk/libs/esl/fs_cli.c Modified: freeswitch/trunk/libs/esl/fs_cli.c ============================================================================== --- freeswitch/trunk/libs/esl/fs_cli.c (original) +++ freeswitch/trunk/libs/esl/fs_cli.c Mon Dec 29 19:31:15 2008 @@ -83,30 +83,26 @@ if (!esl_strlen_zero(type)) { if (!strcasecmp(type, "log/data")) { - int level = 0, tchannel = 0; + int level = 0; const char *lname = esl_event_get_header(handle->last_event, "log-level"); - const char *channel = esl_event_get_header(handle->last_event, "text-channel"); - //const char *file = esl_event_get_header(handle->last_event, "log-file"); - - if (channel) { - tchannel = atoi(channel); - } - +#ifdef WIN32 + DWORD len = (DWORD) strlen(handle->last_event->body); + DWORD outbytes = 0; +#endif if (lname) { level = atoi(lname); } - if (tchannel == 0) { + #ifdef WIN32 - DWORD len = (DWORD) strlen(handle->last_event->body); - DWORD outbytes = 0; - SetConsoleTextAttribute(hStdout, COLORS[level]); - WriteFile(hStdout, handle->last_event->body, len, &outbytes, NULL); - SetConsoleTextAttribute(hStdout, wOldColorAttrs); + + SetConsoleTextAttribute(hStdout, COLORS[level]); + WriteFile(hStdout, handle->last_event->body, len, &outbytes, NULL); + SetConsoleTextAttribute(hStdout, wOldColorAttrs); #else - printf("%s%s%s", COLORS[level], handle->last_event->body, ESL_SEQ_DEFAULT_COLOR); + printf("%s%s%s", COLORS[level], handle->last_event->body, ESL_SEQ_DEFAULT_COLOR); #endif - } + known++; } else if (!strcasecmp(type, "text/disconnect-notice")) { running = thread_running = 0;

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[commit] r10996 - freeswitch/trunk/src
by FreeSWITCH SVN 29 Dec '08

29 Dec '08

Author: anthm Date: Mon Dec 29 18:38:59 2008 New Revision: 10996 Log: dont transmit console replies over the logger engine Modified: freeswitch/trunk/src/switch_console.c Modified: freeswitch/trunk/src/switch_console.c ============================================================================== --- freeswitch/trunk/src/switch_console.c (original) +++ freeswitch/trunk/src/switch_console.c Mon Dec 29 18:38:59 2008 @@ -249,10 +249,18 @@ SWITCH_STANDARD_STREAM(stream); if (stream.data) { + FILE *handle = switch_core_get_console(); + if (switch_api_execute(cmd, arg, NULL, &stream) == SWITCH_STATUS_SUCCESS) { - switch_log_printf(SWITCH_CHANNEL_LOG_CLEAN, SWITCH_LOG_CONSOLE, "API CALL [%s(%s)] output:\n%s\n", cmd, arg ? arg : "", (char *) stream.data); + if (handle) { + fprintf(handle, "API CALL [%s(%s)] output:\n%s\n", cmd, arg ? arg : "", (char *) stream.data); + fflush(handle); + } } else { - switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_CONSOLE, "Unknown Command: %s\n", cmd); + if (handle) { + fprintf(handle, "Unknown Command: %s\n", cmd); + fflush(handle); + } } free(stream.data); } else {

1 0

[commit] r10995 - freeswitch/trunk/debian
by FreeSWITCH SVN 29 Dec '08

29 Dec '08

Author: mcollins Date: Mon Dec 29 18:32:48 2008 New Revision: 10995 Log: Update debian/changelog with 1.0.2 changes Modified: freeswitch/trunk/debian/changelog Modified: freeswitch/trunk/debian/changelog ============================================================================== --- freeswitch/trunk/debian/changelog (original) +++ freeswitch/trunk/debian/changelog Mon Dec 29 18:32:48 2008 @@ -1,3 +1,296 @@ +freeswitch (1.0.2-1) unstable; urgency=low + + * all: don't add module interfaces before returning from error conditions in module load functions (MDXMLINT-36) + * all: fixed multiple memory leaks + * all: improved module unloading/reloading support + * build: add support for --switchconfdir (FSBUILD-84) + * build: fixed netbsd build + * build: make freeswitch stop graceflly with /etc/init.d/freeswitch stop on debian add working dir to start-stop-dir so freeswitch dumps core in workdir + * build: multiple packaging fixes + * build: user freeswitch not added to audio group on deb install (FSBUILD-95) + * Configuration: many updates to default configuration + * core: Add ability to choose uuid from originate string, originate_uuid var (use at your own risk) + * core: add bridge_generate_comfort_noise option for bridge to generate comfort noise to the A leg when there is no audio on the B leg + * core: add chan vars to param event during hangup hook + * core: add exec directive to xml preprocessor (not available on windows) + * core: add force_transfer_dialplan and force_transfer_context variables + * core: add hashing to event header lookup + * core: add hits to tone_detect + * core: add last_dtmf_duration variable + * core: add msleep function to swigged languages + * core: add park_after_bridge variable + * core: add per leg timeouts and specific cause codes in reject_on_single_fail + * core: add runtime selection of the module dir (FSCORE-198) + * core: add scheduler support for heartbeat + * core: add session heartbeat feature + * core: add session.destroy psuedo method to sort of destroy a session at least for the sake of FS + * core: add session.unsetInputCallback + * core: add strftime format string validation for user supplied values + * core: add vars param to switch_ivr_originate for recursion (MODAPP-175) + * core: added a "group" concept to the user directory + * core: added ability to do dns lookup to find ip with host: like stun: (FSCORE-219) + * core: added better locking for codec changes during a call + * core: added current_application and current_application_data variables + * core: added error/ magic endpoint so modules can return error causes in situations like sofia_contact + * core: added read_result channel variable + * core: added support for "F" to indicate flash in dtmf (FSCORE-213) + * core: allow calls to be stolen from originate + * core: allow you to get the privacy bits in the caller_profile + * core: change dso code to load symbols local + * core: changes core flags to be array based so we have more + * core: eavesdrop causes the people being eavesdropped on to not hear ach other (MODAPP-140) + * core: expose time table to variable interface via caller field lookup + * core: fix 100% cpu when sending parked call to moh (FSCORE-234) + * core: fix bridge app to make sure both channels are ready for media when one is only in ringing state + * core: fix buffer overflow (FSCORE-188) + * core: fix conference dial by allowing multiple braces in originate, fix bad pointer op (FSCORE-208) + * core: fix double detection of DTMF in IVR (FSCORE-221) + * core: fix hangup_after_bridge is false on a bridge started with the intercept app + * core: fix issue where pid file is accidentally truncated + * core: fix ivr timeout (FSCORE-181) + * core: fix memory leak in alias tab completion code + * core: fix min digits in read app + * core: fix out-of-bounds pointer in variable expansion (FSCORE-171) + * core: fix segfault in media bugs when in bypass media (FSCORE-193) + * core: fix segfault on gtalk to sip calls (FSCORE-212) + * core: fix segfault on reloadxml (FSCORE-176) + * core: fix segfault on trasfering eavesdopping party (FSCORE-210) + * core: fix segfault using switch_system function (FSCORE-196) + * core: fix session.bridge + * core: fix setting effective_caller_id_name / effective_caller_id_number on bridge dialstring (MODAPP-122) + * core: fix stream_raw_write (MODAPP-145) + * core: fix using resampling on ringback file + * core: fixed performance bottleneck in sqlite db's + * core: fixed race in reloadxml + * core: increment app before execute in case it returns to execute it will go to the next item in the list and not the same + * core: ivr menu max_failures and max_timeouts now default to 3 if not specified or invalid (less than 1) values are specified (FSCORE-244) + * core: ivr_menu max-timeouts option, result in ivr_menu_status var (FSCORE-183) + * core: let b legs use park_after_bridge too + * core: make events less verbose unless verbose_events is set + * core: parse private events during originate + * core: pass pdd data to a leg on oubound calls using bridge + * core: prevent crash in crazy situation with xml interface lookup failures (FSCORE-169) + * core: reduce cpu requirement for generated comfort noise + * core: remove interface names from core db on unload + * core: reworked timing resulting in significant performance increase and better rtp timing + * core: rewrite switch_play_and_get_digits (MODAPP-166) + * core: session.recordFile never terminates (MODLANG-79) + * core: session.transfer make dialplan and context optional + * core: set_user app now sets domain vars as well as user vars + * core: tone_detect not triggering app after tone detection (MODAPP-182) + * core: unprivileged user setting bigger stack for switch_system thread failure (FSCORE-197) + * core: user_exists returns false when fetching a user from XML Curl or other xml interfaces + * libesl: added c event socket library and fs_cli + * libsndfile: fix autoconf 2.62 support (LBSNDF-5) + * mod commands: add "all" modifier to "break" command + * mod_celt: added new module + * mod_commands: Add support for more than 2 variables to uuid_setvar_multi (MODAPP-171) + * mod_commands: Add support for passing the cause of hangup to the uuid_kill command (FSCORE-217) + * mod_commands: add attr lookup to user_data + * mod_commands: add domain_exists fsapi command + * mod_commands: add eval fsapi command + * mod_commands: add flush_dtmf app and uuid_flush_dtmf api command + * mod_commands: add fsctl send_sighup, fsctl shutdown asap, unsched_api commands + * mod_commands: add fsctl shutdown [elegant|restart|cancel] + * mod_commands: add new syntax to uuid_setvar to allow you to unset a var. <uuid> <var> [value] (MODAPP-167) + * mod_commands: add reload fsapi command to reload a module + * mod_commands: add system fsapi and application (MODAPP-138) + * mod_commands: added hupall fsapi command + * mod_commands: added strftime_tz api command + * mod_commands: break all now stops broadcast too + * mod_commands: fix api command sent through sched_api was getting the last char lopped off + * mod_commands: fix race on transfer with -both + * mod_commands: fix system dialplan app problems (MODAPP-86) + * mod_commands: only send content-type on status when it really is http. + * mod_conference: add fsapi to stop async playback too + * mod_conference: add video caps to mod_conference with video follow audio + * mod_conference: better sound prefix handling when using say: and allow say: on kick sounds. + * mod_conference: fix race in record + * mod_conference: fix runaway thread when floor holder has no video and other people do have video + * mod_conference: fix seg when kicking many members quickly (MODAPP-129) + * mod_conference: fix segfault on invalid chat event + * mod_conference: perpetual sound does not auto-mute, you can do that yourself if you want it + * mod_dialplan_xml: add Hunt- vars in dialplan lookup after transfer + * mod_dialplan_xml: fail call on extensions with nested conditions + * mod_dingaling: (LBDING-7) fix segfault on os x + * mod_dingaling: end call on ice timeout + * mod_dingaling: fix presence on jabber to be less protocol ambiguous + * mod_dingaling: fix segfault (LBDING-10) + * mod_dingaling: update to support latest client from google + * mod_dptools: add a mechanism to tell if a file played from sendmsg over event socket + * mod_dptools: add playback_terminator support to phrase and say app + * mod_dptools: add playback_terminator_used variable (MODAPP-132) + * mod_dptools: add presence application + * mod_dptools: fix originate api not parsing users properly (FSCORE-246) + * mod_dptools: fix record and record_session to create directory if it does not exist (FSCORE-250) + * mod_dptools: fixed limit and + parsing bug in record_session app (MODAPP-148) + * mod_dptools: remove_bugs added to remove all media bugs on a session + * mod_erlang_event: add new module + * mod_event_socket: missing : after Content-Length in event socket (MODEVENT-33) + * mod_event_socket: add event socket listener filters + * mod_event_socket: add stateful listener fsapi commands for ajax-y type event interface over http + * mod_event_socket: fix arg parsing errors (MODEVENT-34) + * mod_event_socket: fix shutdown segfault race (MODEVENT-32) + * mod_event_socket: inbound connection to event_socket can now take over an existing channel with 'myevents <uuid>' to take on the behaviour of an outbound socket + * mod_event_socket: let any channel get messages + * mod_event_socket: make event socket wait for hangup on outbound mode and send disconnect message + * mod_expr: fix endless loop + * mod_fax: new module + * mod_fifo: add fifo_consumer_wrapup_time var (MODAPP-117) + * mod_fifo: added callback agents + * mod_fifo: honor keyword silence (MODAPP-118) + * mod_flite: added windows build + * mod_fsv: fix in a windows enviroment opening the record file in text mode. (MODAPP-169) + * mod_http: added new module + * mod_java: updated to new module api to support read/write locks on interface + * mod_limit: accept dialplan context for transfer (MODAPP-161) + * mod_limit: added hashtable based limit functions + * mod_limit: prevent empty error log message (MODAPP-134) + * mod_local_stream: add start_local_stream and stop_local_stream fsapi commands to start/stop dynamically (MODFORM-13) + * mod_local_stream: fix leak and improve error checking + * mod_local_stream: fix seg when no timer name specified in config file. (MODFORM-16) + * mod_loopback: add new module + * mod_lua: add local scripts directory support (MODLANG-86) + * mod_lua: don't eval blank string + * mod_lua: fix originate + * mod_lua: fix segfault (MODLANG-77) + * mod_lua: update to lua 5.1.4 (MODLANG-87) + * mod_lumenvox: removed + * mod_managed: new module replaces mod_mono now supports native .net runtime on windows as well + * mod_opal: added to trunk (still very beta) + * mod_perl: fix segfault (MODLANG-77) + * mod_pocketsphinx: fix rpm build + * mod_portaudio: fix cpu race on inbound call to pa when no ring file is set + * mod_radius_cdr: dictionary update for cause code changes (MODEVENT-27) + * mod_radius_cdr: fix unload (MODEVENT-29) + * mod_shout: add stereo recording broadcast support + * mod_shout: added windows build + * mod_shout: fix segfault when recording mp3's (MODFORM-12) + * mod_shout: improved stability of mp3 decoding + * mod_siren: added new module + * mod_sndfile added support to record 16bit for the various rates including 48kHz + * mod_sofia: Add filter to "sofia status profile XXX" (MODENDP-138) + * mod_sofia: Add force-register-db-domain which works in conjunction with force-register-domain. + * mod_sofia: Add optional <variables> and <params> tag to <gateway> tag. + * mod_sofia: Challenge the right realm when to_host is outside the users domain. (MODENDP-136) + * mod_sofia: Improve notify messages through a proxy (MODENDP-147) + * mod_sofia: MWI for multiple domains (MODAPP-126) + * mod_sofia: Move "a=sendrecv" from session to media section of SDP (MODENDP-148) + * mod_sofia: add 200 OK re-invite without sdp + * mod_sofia: add custom sofia::gateway_state event (MODENDP-112) + * mod_sofia: add fire events for the refer SIP NOTIFY event package (MODENDP-152) + * mod_sofia: add more params for xml_curl directory lookup + * mod_sofia: add new auto vals for challenge-realm param <param name="challenge-realm" value="auto_from|auto_to|<hardcoded_val>"/> + * mod_sofia: add option to turn of auto_restart of sofia profiles on ip change + * mod_sofia: add params to use sip callid as uuid on inbound calls and uuid as sip callid on outbound calls + * mod_sofia: add parsing of Privacy header for privacy info (MODENDP-133) + * mod_sofia: add proto_specific_hangup_cause to both legs + * mod_sofia: add proxy 3pcc mode + * mod_sofia: add redirect variable to channel as well as partner channe (MODENDP-135) + * mod_sofia: add sip-forbid-register to user params to refuse to let a certian user register + * mod_sofia: add sip: into register-proxy when it's not specified + * mod_sofia: add sip_history_info var for inbound invites. + * mod_sofia: add sip_via_protocol variable + * mod_sofia: add sofia xmlstatus (MODENDP-156) + * mod_sofia: add support for params other than Replaces in Refer-To (MODENDP-143) + * mod_sofia: add support for profiles sharing databases so that you can have a domain that uses multiple profiles for split dns type setups + * mod_sofia: add support for refer transfer involving multiple machines + * mod_sofia: add support to send a notify in the invite dialog by specifying the uuid of the call. (SFSIP-92) + * mod_sofia: add suppress_from_cidname var to not have display name in from header (MODENDP-153) + * mod_sofia: added sip_hangup_disposition variable + * mod_sofia: allow send_message and notify events to send a message/notify without a body if needed. + * mod_sofia: append -1 .. -N postfix after any X-headers as vars that have the same name + * mod_sofia: cache auth_gateway_name in sofia for challenged bye + * mod_sofia: cancel proxy or no-media mode if you purposely answer or pre_answer + * mod_sofia: correct result code mapping for Unallocated Number (MODENDP-124) + * mod_sofia: disable 100rel by default + * mod_sofia: don't accept crypto in the RTP/AVP (MODENDP-126) + * mod_sofia: don't put CN in sdp answer if it was not in the offer. + * mod_sofia: fix Incorrect IP address shows up in SDP "o" field when multiple external IPs available and FS not bound to first (MODENDP-132) + * mod_sofia: fix Wrong RTP media port destination after reinvite/UNHOLD (SFSIP-82) + * mod_sofia: fix bug on linksys where they lie about the ptime and handle linksys transfer problem + * mod_sofia: fix chat (send an IM) assumes that the user's profile is the same as their domain, which isn't necessarily so (SFSIP-83) + * mod_sofia: fix dtmf handling of broken info dtmf endpoints + * mod_sofia: fix eyebeam presence to be RFC compliant (MODENDP-144) + * mod_sofia: fix ip change detection when in proxy mode + * mod_sofia: fix register_proxy ignoring the paramaters (MODENDP-121) + * mod_sofia: fix remote session refresh triggers request glare (MODENDP-131) + * mod_sofia: fix rtp auto adjust running when it should not + * mod_sofia: fix rtp sent to wrong port after some re-INVITE scenarios (MODENDP-141) + * mod_sofia: fix sending of cn packets across bridge when we shouldn't + * mod_sofia: fix sqlite issue with select of the sip contact + * mod_sofia: fixed segfault on invalid presence payload + * mod_sofia: gateway ping needs to look for 501 (SFSIP-78) + * mod_sofia: handle multi contact register responses and register timeout better + * mod_sofia: improve gateway resilience + * mod_sofia: log ip and port you get reply to invite from + * mod_sofia: make multiple-registations=true use the contact method and call-id option to do it the old way + * mod_sofia: make proxy mode pull the port from m=image as well + * mod_sofia: make register-proxy preserve the url composed from proxy but target the packets to desired address (MODENDP-121) + * mod_sofia: many fixes for sonus rtp issues silence_when_idle=400 chanvar to send generated silence duing sleeps etc + * mod_sofia: many fixes in presence handling + * mod_sofia: passthrough t.38 fixes + * mod_sofia: pick ipv4 or ipv6 based on sipip instead of having mixed in sdp + * mod_sofia: send NOTIFY on TCP/UDP depending on the SUBSCRIBE (SFSIP-104) + * mod_sofia: setting profile option multiple-registrations=contact key multi reg off the contact string + * mod_sofia: wait for a reply on refer + * mod_soundtouch: fixes and improvements, many options changed (MODAPP-149) + * mod_soundtouch: updated to new module api + * mod_spidermonkey: Segmentation fault in check_hangup_hook at mod_spidermonkey.c:1589 (MODLANG-74) + * mod_spidermonkey: fix bug in apiExecute + * mod_spidermonkey: fix memory pool handling and leaks + * mod_spidermonkey: limit recursion busting through the stack (FSCORE-202) + * mod_spidermonkey: make session.getVariable return blank string not the word false + * mod_spidermonkey_curl: add optional content-type arg + * mod_spidermonkey_odbc: fix numRows and add numCols + * mod_spidermonkey_odbc: fix segfault (MODLANG-75) + * mod_stress: new module for voice stress analysis + * mod_syslog: don't log blank lines (FSCORE-163) + * mod_tone_stream: let silence_stream://0 indicate perpetual silence + * mod_vmd: add new module to detect voicemail "beep" + * mod_voicemail: Add vm_alternate_greet_id param to directory entry (MODAPP-174) + * mod_voicemail: Patch to add voicemail preference controlling date announcement new param 'play-date-announcement' to values 'first' 'last' or 'never' defaults to first to retain previous behavior (MODAPP-121) + * mod_voicemail: Update mwi light after delete vm via web. (MODAPP-124) + * mod_voicemail: add ability to get to options without listening to every saved message (MODAPP-115) + * mod_voicemail: add ability to skip greeting when leaving a voicemail. (MODAPP-181) + * mod_voicemail: add change-pass-key config file option + * mod_voicemail: add forwarding support + * mod_voicemail: add local dtmf driven alternat vm pass + * mod_voicemail: add proper notification of a vm message being too short + * mod_voicemail: add support for auth via a1-hash + * mod_voicemail: add the "storage-dir" parameter to be set on a per-user basis (MODAPP-133) + * mod_voicemail: add voicemail_greeting_path variable + * mod_voicemail: added voicemail_alternate_greet_id variable + * mod_voicemail: allow changing of password from voicemail to update user directory if using non-static config (MODAPP-156) + * mod_voicemail: created email date (int overflow) (MODAPP-125) + * mod_voicemail: don't try to deliver vm when no file was recorded. (MODAPP-133) + * mod_voicemail: fix MWI with xml_curl used for directory (MODAPP-176) + * mod_voicemail: fix Voicemail messages occasionally lost / stranded (MODAPP-178) + * mod_voicemail: fix invalid event after message deleted (MODAPP-170) + * mod_voicemail: fix mwi for phones with multiple registrations problem (MODAPP-153) + * mod_voicemail: fix voicemail segfault on incorrect password (FSCORE-187) + * mod_voicemail: fix voicemail_inject error handling (MODAPP-133) + * mod_voicemail: fix voicemail_inject usage api call + * mod_voicemail: improve error checking (MODAPP-142) + * mod_voicemail: localize notification emails (MODAPP-139) + * mod_voicemail: make more multi-domain friendly (MODAPP-162) + * mod_voicemail: make playback created file macros optional (MODAPP-150) + * mod_voicemail: recognize operator key in more places (MODAPP-159) + * mod_voicemail: web interface displays incorrect created / last heard dates (MODAPP-123) + * mod_wanpipe: removed + * mod_xml_cdr: add https support + * mod_xml_cdr: add optional a-leg prefix to xml cdr filenames (MDXMLINT-39) + * mod_xml_cdr: add support for fallback webserver for cdr posting (FSCORE-238) + * mod_xml_curl: Allow specification of HTTP method, and dynamic expansion of variables in URI. (MDXMLINT-41) + * mod_xml_curl: added redirect following (max 10) + * mod_xml_ldap: almost a complete rewrite of this module + * mod_xml_rpc: allow setting of global realm without a global user (MDXMLINT-45) + * mod_xml_rpc: fix multiple segfaults + * mod_xml_rpc: fix segfault on originate via http + * sofia-sip: updated to 1.12.10 (plus a few patches) + + -- Mike Jerris <mike(a)jerris.com> Mon, 29 Dec 2008 14:46:00 -0500 + freeswitch (1.0.1-1) unstable; urgency=low * FIX: prevent intercept race condition that can also be solved with continue_on_fail=originator_cancel @@ -295,7 +588,7 @@ Fixed mod_iax race conditions Fixed ptime negotiation issues when re-packetizing Added ip based acl lists - + * -- Michael Jerris <mike(a)jerris.com> Wed, 9 Apr 2008 12:58:22 -0400 freeswitch (1.0~rc1-1) unstable; urgency=low

1 0

[commit] r10994 - freeswitch/trunk/src
by FreeSWITCH SVN 29 Dec '08

29 Dec '08

Author: anthm Date: Mon Dec 29 17:41:03 2008 New Revision: 10994 Log: fix FSCORE-261 Modified: freeswitch/trunk/src/switch_core.c Modified: freeswitch/trunk/src/switch_core.c ============================================================================== --- freeswitch/trunk/src/switch_core.c (original) +++ freeswitch/trunk/src/switch_core.c Mon Dec 29 17:41:03 2008 @@ -168,7 +168,7 @@ SWITCH_DECLARE(void) switch_core_remove_state_handler(const switch_state_handler_table_t *state_handler) { - int index, total = 0; + int index, tmp_index = 0; const switch_state_handler_table_t *tmp[SWITCH_MAX_STATE_HANDLERS+1] = { 0 }; switch_mutex_lock(runtime.global_mutex); @@ -179,13 +179,12 @@ if (cur == state_handler) { continue; } - tmp[index] = runtime.state_handlers[index]; - total++; + tmp[tmp_index++] = cur; } runtime.state_handler_index = 0; - for (index = 0; index < total; index++) { + for (index = 0; index < tmp_index; index++) { runtime.state_handlers[runtime.state_handler_index++] = tmp[index]; } switch_mutex_unlock(runtime.global_mutex);

1 0

[commit] r10993 - in freeswitch/trunk/src: . include mod/applications/mod_voicemail
by FreeSWITCH SVN 29 Dec '08

29 Dec '08

Author: anthm Date: Mon Dec 29 17:26:10 2008 New Revision: 10993 Log: fix voicemail to be in line with groups Modified: freeswitch/trunk/src/include/switch_xml.h freeswitch/trunk/src/mod/applications/mod_voicemail/mod_voicemail.c freeswitch/trunk/src/switch_xml.c Modified: freeswitch/trunk/src/include/switch_xml.h ============================================================================== --- freeswitch/trunk/src/include/switch_xml.h (original) +++ freeswitch/trunk/src/include/switch_xml.h Mon Dec 29 17:26:10 2008 @@ -350,6 +350,8 @@ switch_xml_t *root, switch_xml_t *domain, switch_xml_t *user, switch_xml_t *ingroup, switch_event_t *params); +SWITCH_DECLARE(switch_status_t) switch_xml_locate_user_in_domain(const char *user_name, switch_xml_t domain, switch_xml_t *user, switch_xml_t *ingroup); + ///\brief open a config in the core registry ///\param file_path the name of the config section e.g. modules.conf ///\param node a pointer to point to the node if it is found Modified: freeswitch/trunk/src/mod/applications/mod_voicemail/mod_voicemail.c ============================================================================== --- freeswitch/trunk/src/mod/applications/mod_voicemail/mod_voicemail.c (original) +++ freeswitch/trunk/src/mod/applications/mod_voicemail/mod_voicemail.c Mon Dec 29 17:26:10 2008 @@ -2261,7 +2261,9 @@ } /*switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "Send mail is %d, var is %s\n", send_mail, var);*/ } - switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "Send mail is %d and that's my final answer\n", send_mail); + + switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "Deliver VM to %s@%s\n", myid, domain_name); + if (!switch_strlen_zero(vm_storage_dir)) { dir_path = switch_mprintf("%s%s%s", vm_storage_dir, SWITCH_PATH_SEPARATOR, myid); } else if (!switch_strlen_zero(profile->storage_dir)) { @@ -2507,7 +2509,7 @@ vm_profile_t *profile; char *dup = NULL, *user = NULL, *domain = NULL; switch_status_t status = SWITCH_STATUS_SUCCESS; - int istag = 0, isall = 0; + int isgroup = 0, isall = 0; int argc = 0; char *argv[4]; char *box, *path, *cid_num, *cid_name; @@ -2540,9 +2542,9 @@ domain = user; } - if (switch_stristr("tag=", user)) { - user += 4; - istag++; + if (switch_stristr("group=", user)) { + user += 6; + isgroup++; } else if (user == domain) { isall++; } @@ -2567,6 +2569,17 @@ switch_event_create(&my_params, SWITCH_EVENT_REQUEST_PARAMS); switch_assert(my_params); + + if (isgroup) { + switch_event_add_header_string(my_params, SWITCH_STACK_BOTTOM, "group", user); + } else { + if (isall) { + switch_event_add_header_string(my_params, SWITCH_STACK_BOTTOM, "user", "_all_"); + } else { + switch_event_add_header_string(my_params, SWITCH_STACK_BOTTOM, "user", user); + } + } + switch_event_add_header_string(my_params, SWITCH_STACK_BOTTOM, "domain", domain); switch_event_add_header_string(my_params, SWITCH_STACK_BOTTOM, "purpose", "publish-vm"); @@ -2581,24 +2594,66 @@ switch_core_new_memory_pool(&pool); - if (!isall && !istag) { - if ((ut = switch_xml_find_child(x_domain, "user", "id", user))) { + + if (isgroup) { + switch_xml_t group = NULL, groups = NULL, users = NULL; + if ((groups = switch_xml_child(x_domain, "groups"))) { + if ((group = switch_xml_find_child_multi(groups, "group", "name", user, NULL))) { + if ((users = switch_xml_child(group, "users"))) { + for (ut = switch_xml_child(users, "user"); ut; ut = ut->next) { + const char *type = switch_xml_attr_soft(ut, "type"); + + if (!strcasecmp(type, "pointer")) { + const char *uname = switch_xml_attr_soft(ut, "id"); + switch_xml_t ux; + + if (switch_xml_locate_user_in_domain(uname, x_domain, &ux, NULL) == SWITCH_STATUS_SUCCESS) { + switch_event_create(&my_params, SWITCH_EVENT_REQUEST_PARAMS); + status = deliver_vm(profile, ux, domain, path, 0, "B", my_params, pool, cid_name, cid_num, SWITCH_TRUE); + switch_event_destroy(&my_params); + } + continue; + } + + switch_event_create(&my_params, SWITCH_EVENT_REQUEST_PARAMS); + status = deliver_vm(profile, ut, domain, path, 0, "B", my_params, pool, cid_name, cid_num, SWITCH_TRUE); + switch_event_destroy(&my_params); + } + } + } else { + switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_WARNING, "Cannot locate group %s\n", user); + } + } + } else if (isall) { + switch_xml_t group = NULL, groups = NULL, users = NULL; + if ((groups = switch_xml_child(x_domain, "groups"))) { + for (group = switch_xml_child(groups, "group"); group; group = group->next) { + if ((users = switch_xml_child(group, "users"))) { + for (ut = switch_xml_child(users, "user"); ut; ut = ut->next) { + const char *type = switch_xml_attr_soft(ut, "type"); + + if (!strcasecmp(type, "pointer")) { + continue; + } + + switch_event_create(&my_params, SWITCH_EVENT_REQUEST_PARAMS); + status = deliver_vm(profile, ut, domain, path, 0, "B", my_params, pool, cid_name, cid_num, SWITCH_TRUE); + switch_event_destroy(&my_params); + } + } + } + } + + } else { + switch_xml_t x_group = NULL; + + if ((status = switch_xml_locate_user_in_domain(user, x_domain, &ut, &x_group)) == SWITCH_STATUS_SUCCESS) { switch_event_create(&my_params, SWITCH_EVENT_REQUEST_PARAMS); status = deliver_vm(profile, ut, domain, path, 0, "B", my_params, pool, cid_name, cid_num, SWITCH_TRUE); switch_event_destroy(&my_params); } else { status = SWITCH_STATUS_FALSE; } - } else { - for (ut = switch_xml_child(x_domain, "user"); ut; ut = ut->next) { - const char *tag; - - if (isall || (istag && (tag=switch_xml_attr(ut, "vm-tag")) && !strcasecmp(tag, user))) { - switch_event_create(&my_params, SWITCH_EVENT_REQUEST_PARAMS); - status = deliver_vm(profile, ut, domain, path, 0, "B", my_params, pool, cid_name, cid_num, SWITCH_TRUE); - switch_event_destroy(&my_params); - } - } } switch_core_destroy_memory_pool(&pool); @@ -3562,7 +3617,7 @@ } } -#define VM_INJECT_USAGE "[tag=]<box> <sound_file> [<cid_num>] [<cid_name>]" +#define VM_INJECT_USAGE "[group=]<box> <sound_file> [<cid_num>] [<cid_name>]" SWITCH_STANDARD_API(voicemail_inject_api_function) { if (voicemail_inject(cmd) == SWITCH_STATUS_SUCCESS) { Modified: freeswitch/trunk/src/switch_xml.c ============================================================================== --- freeswitch/trunk/src/switch_xml.c (original) +++ freeswitch/trunk/src/switch_xml.c Mon Dec 29 17:26:10 2008 @@ -1634,7 +1634,6 @@ return status; } - static switch_status_t find_user_in_tag(switch_xml_t tag, const char *ip, const char *user_name, const char *key, switch_event_t *params, switch_xml_t *user) { const char *type = "!pointer"; @@ -1670,6 +1669,26 @@ } +SWITCH_DECLARE(switch_status_t) switch_xml_locate_user_in_domain(const char *user_name, switch_xml_t domain, switch_xml_t *user, switch_xml_t *ingroup) +{ + switch_xml_t group = NULL, groups = NULL, users = NULL; + switch_status_t status = SWITCH_STATUS_FALSE; + + if ((groups = switch_xml_child(domain, "groups"))) { + for (group = switch_xml_child(groups, "group"); group; group = group->next) { + if ((users = switch_xml_child(group, "users"))) { + if ((status = find_user_in_tag(users, NULL, user_name, "id", NULL, user)) == SWITCH_STATUS_SUCCESS) { + if (ingroup) { + *ingroup = group; + } + break; + } + } + } + } + + return status; +} SWITCH_DECLARE(switch_status_t) switch_xml_locate_user(const char *key, const char *user_name,

1 0

[commit] r10992 - freeswitch/trunk/docs
by FreeSWITCH SVN 29 Dec '08

29 Dec '08

Author: mikej Date: Mon Dec 29 17:19:09 2008 New Revision: 10992 Log: Change-log Modified: freeswitch/trunk/docs/ChangeLog Modified: freeswitch/trunk/docs/ChangeLog ============================================================================== --- freeswitch/trunk/docs/ChangeLog (original) +++ freeswitch/trunk/docs/ChangeLog Mon Dec 29 17:19:09 2008 @@ -1,3 +1,294 @@ +freeswitch (1.0.2) + + all: don't add module interfaces before returning from error conditions in module load functions (MDXMLINT-36) + all: fixed multiple memory leaks + all: improved module unloading/reloading support + build: add support for --switchconfdir (FSBUILD-84) + build: fixed netbsd build + build: make freeswitch stop graceflly with /etc/init.d/freeswitch stop on debian add working dir to start-stop-dir so freeswitch dumps core in workdir + build: multiple packaging fixes + build: user freeswitch not added to audio group on deb install (FSBUILD-95) + Configuration: many updates to default configuration + core: Add ability to choose uuid from originate string, originate_uuid var (use at your own risk) + core: add bridge_generate_comfort_noise option for bridge to generate comfort noise to the A leg when there is no audio on the B leg + core: add chan vars to param event during hangup hook + core: add exec directive to xml preprocessor (not available on windows) + core: add force_transfer_dialplan and force_transfer_context variables + core: add hashing to event header lookup + core: add hits to tone_detect + core: add last_dtmf_duration variable + core: add msleep function to swigged languages + core: add park_after_bridge variable + core: add per leg timeouts and specific cause codes in reject_on_single_fail + core: add runtime selection of the module dir (FSCORE-198) + core: add scheduler support for heartbeat + core: add session heartbeat feature + core: add session.destroy psuedo method to sort of destroy a session at least for the sake of FS + core: add session.unsetInputCallback + core: add strftime format string validation for user supplied values + core: add vars param to switch_ivr_originate for recursion (MODAPP-175) + core: added a "group" concept to the user directory + core: added ability to do dns lookup to find ip with host: like stun: (FSCORE-219) + core: added better locking for codec changes during a call + core: added current_application and current_application_data variables + core: added error/ magic endpoint so modules can return error causes in situations like sofia_contact + core: added read_result channel variable + core: added support for "F" to indicate flash in dtmf (FSCORE-213) + core: allow calls to be stolen from originate + core: allow you to get the privacy bits in the caller_profile + core: change dso code to load symbols local + core: changes core flags to be array based so we have more + core: eavesdrop causes the people being eavesdropped on to not hear ach other (MODAPP-140) + core: expose time table to variable interface via caller field lookup + core: fix 100% cpu when sending parked call to moh (FSCORE-234) + core: fix bridge app to make sure both channels are ready for media when one is only in ringing state + core: fix buffer overflow (FSCORE-188) + core: fix conference dial by allowing multiple braces in originate, fix bad pointer op (FSCORE-208) + core: fix double detection of DTMF in IVR (FSCORE-221) + core: fix hangup_after_bridge is false on a bridge started with the intercept app + core: fix issue where pid file is accidentally truncated + core: fix ivr timeout (FSCORE-181) + core: fix memory leak in alias tab completion code + core: fix min digits in read app + core: fix out-of-bounds pointer in variable expansion (FSCORE-171) + core: fix segfault in media bugs when in bypass media (FSCORE-193) + core: fix segfault on gtalk to sip calls (FSCORE-212) + core: fix segfault on reloadxml (FSCORE-176) + core: fix segfault on trasfering eavesdopping party (FSCORE-210) + core: fix segfault using switch_system function (FSCORE-196) + core: fix session.bridge + core: fix setting effective_caller_id_name / effective_caller_id_number on bridge dialstring (MODAPP-122) + core: fix stream_raw_write (MODAPP-145) + core: fix using resampling on ringback file + core: fixed performance bottleneck in sqlite db's + core: fixed race in reloadxml + core: increment app before execute in case it returns to execute it will go to the next item in the list and not the same + core: ivr menu max_failures and max_timeouts now default to 3 if not specified or invalid (less than 1) values are specified (FSCORE-244) + core: ivr_menu max-timeouts option, result in ivr_menu_status var (FSCORE-183) + core: let b legs use park_after_bridge too + core: make events less verbose unless verbose_events is set + core: parse private events during originate + core: pass pdd data to a leg on oubound calls using bridge + core: prevent crash in crazy situation with xml interface lookup failures (FSCORE-169) + core: reduce cpu requirement for generated comfort noise + core: remove interface names from core db on unload + core: reworked timing resulting in significant performance increase and better rtp timing + core: rewrite switch_play_and_get_digits (MODAPP-166) + core: session.recordFile never terminates (MODLANG-79) + core: session.transfer make dialplan and context optional + core: set_user app now sets domain vars as well as user vars + core: tone_detect not triggering app after tone detection (MODAPP-182) + core: unprivileged user setting bigger stack for switch_system thread failure (FSCORE-197) + core: user_exists returns false when fetching a user from XML Curl or other xml interfaces + libesl: added c event socket library and fs_cli + libsndfile: fix autoconf 2.62 support (LBSNDF-5) + mod commands: add "all" modifier to "break" command + mod_celt: added new module + mod_commands: Add support for more than 2 variables to uuid_setvar_multi (MODAPP-171) + mod_commands: Add support for passing the cause of hangup to the uuid_kill command (FSCORE-217) + mod_commands: add attr lookup to user_data + mod_commands: add domain_exists fsapi command + mod_commands: add eval fsapi command + mod_commands: add flush_dtmf app and uuid_flush_dtmf api command + mod_commands: add fsctl send_sighup, fsctl shutdown asap, unsched_api commands + mod_commands: add fsctl shutdown [elegant|restart|cancel] + mod_commands: add new syntax to uuid_setvar to allow you to unset a var. <uuid> <var> [value] (MODAPP-167) + mod_commands: add reload fsapi command to reload a module + mod_commands: add system fsapi and application (MODAPP-138) + mod_commands: added hupall fsapi command + mod_commands: added strftime_tz api command + mod_commands: break all now stops broadcast too + mod_commands: fix api command sent through sched_api was getting the last char lopped off + mod_commands: fix race on transfer with -both + mod_commands: fix system dialplan app problems (MODAPP-86) + mod_commands: only send content-type on status when it really is http. + mod_conference: add fsapi to stop async playback too + mod_conference: add video caps to mod_conference with video follow audio + mod_conference: better sound prefix handling when using say: and allow say: on kick sounds. + mod_conference: fix race in record + mod_conference: fix runaway thread when floor holder has no video and other people do have video + mod_conference: fix seg when kicking many members quickly (MODAPP-129) + mod_conference: fix segfault on invalid chat event + mod_conference: perpetual sound does not auto-mute, you can do that yourself if you want it + mod_dialplan_xml: add Hunt- vars in dialplan lookup after transfer + mod_dialplan_xml: fail call on extensions with nested conditions + mod_dingaling: (LBDING-7) fix segfault on os x + mod_dingaling: end call on ice timeout + mod_dingaling: fix presence on jabber to be less protocol ambiguous + mod_dingaling: fix segfault (LBDING-10) + mod_dingaling: update to support latest client from google + mod_dptools: add a mechanism to tell if a file played from sendmsg over event socket + mod_dptools: add playback_terminator support to phrase and say app + mod_dptools: add playback_terminator_used variable (MODAPP-132) + mod_dptools: add presence application + mod_dptools: fix originate api not parsing users properly (FSCORE-246) + mod_dptools: fix record and record_session to create directory if it does not exist (FSCORE-250) + mod_dptools: fixed limit and + parsing bug in record_session app (MODAPP-148) + mod_dptools: remove_bugs added to remove all media bugs on a session + mod_erlang_event: add new module + mod_event_socket: missing : after Content-Length in event socket (MODEVENT-33) + mod_event_socket: add event socket listener filters + mod_event_socket: add stateful listener fsapi commands for ajax-y type event interface over http + mod_event_socket: fix arg parsing errors (MODEVENT-34) + mod_event_socket: fix shutdown segfault race (MODEVENT-32) + mod_event_socket: inbound connection to event_socket can now take over an existing channel with 'myevents <uuid>' to take on the behaviour of an outbound socket + mod_event_socket: let any channel get messages + mod_event_socket: make event socket wait for hangup on outbound mode and send disconnect message + mod_expr: fix endless loop + mod_fax: new module + mod_fifo: add fifo_consumer_wrapup_time var (MODAPP-117) + mod_fifo: added callback agents + mod_fifo: honor keyword silence (MODAPP-118) + mod_flite: added windows build + mod_fsv: fix in a windows enviroment opening the record file in text mode. (MODAPP-169) + mod_http: added new module + mod_java: updated to new module api to support read/write locks on interface + mod_limit: accept dialplan context for transfer (MODAPP-161) + mod_limit: added hashtable based limit functions + mod_limit: prevent empty error log message (MODAPP-134) + mod_local_stream: add start_local_stream and stop_local_stream fsapi commands to start/stop dynamically (MODFORM-13) + mod_local_stream: fix leak and improve error checking + mod_local_stream: fix seg when no timer name specified in config file. (MODFORM-16) + mod_loopback: add new module + mod_lua: add local scripts directory support (MODLANG-86) + mod_lua: don't eval blank string + mod_lua: fix originate + mod_lua: fix segfault (MODLANG-77) + mod_lua: update to lua 5.1.4 (MODLANG-87) + mod_lumenvox: removed + mod_managed: new module replaces mod_mono now supports native .net runtime on windows as well + mod_opal: added to trunk (still very beta) + mod_perl: fix segfault (MODLANG-77) + mod_pocketsphinx: fix rpm build + mod_portaudio: fix cpu race on inbound call to pa when no ring file is set + mod_radius_cdr: dictionary update for cause code changes (MODEVENT-27) + mod_radius_cdr: fix unload (MODEVENT-29) + mod_shout: add stereo recording broadcast support + mod_shout: added windows build + mod_shout: fix segfault when recording mp3's (MODFORM-12) + mod_shout: improved stability of mp3 decoding + mod_siren: added new module + mod_sndfile added support to record 16bit for the various rates including 48kHz + mod_sofia: Add filter to "sofia status profile XXX" (MODENDP-138) + mod_sofia: Add force-register-db-domain which works in conjunction with force-register-domain. + mod_sofia: Add optional <variables> and <params> tag to <gateway> tag. + mod_sofia: Challenge the right realm when to_host is outside the users domain. (MODENDP-136) + mod_sofia: Improve notify messages through a proxy (MODENDP-147) + mod_sofia: MWI for multiple domains (MODAPP-126) + mod_sofia: Move "a=sendrecv" from session to media section of SDP (MODENDP-148) + mod_sofia: add 200 OK re-invite without sdp + mod_sofia: add custom sofia::gateway_state event (MODENDP-112) + mod_sofia: add fire events for the refer SIP NOTIFY event package (MODENDP-152) + mod_sofia: add more params for xml_curl directory lookup + mod_sofia: add new auto vals for challenge-realm param <param name="challenge-realm" value="auto_from|auto_to|<hardcoded_val>"/> + mod_sofia: add option to turn of auto_restart of sofia profiles on ip change + mod_sofia: add params to use sip callid as uuid on inbound calls and uuid as sip callid on outbound calls + mod_sofia: add parsing of Privacy header for privacy info (MODENDP-133) + mod_sofia: add proto_specific_hangup_cause to both legs + mod_sofia: add proxy 3pcc mode + mod_sofia: add redirect variable to channel as well as partner channe (MODENDP-135) + mod_sofia: add sip-forbid-register to user params to refuse to let a certian user register + mod_sofia: add sip: into register-proxy when it's not specified + mod_sofia: add sip_history_info var for inbound invites. + mod_sofia: add sip_via_protocol variable + mod_sofia: add sofia xmlstatus (MODENDP-156) + mod_sofia: add support for params other than Replaces in Refer-To (MODENDP-143) + mod_sofia: add support for profiles sharing databases so that you can have a domain that uses multiple profiles for split dns type setups + mod_sofia: add support for refer transfer involving multiple machines + mod_sofia: add support to send a notify in the invite dialog by specifying the uuid of the call. (SFSIP-92) + mod_sofia: add suppress_from_cidname var to not have display name in from header (MODENDP-153) + mod_sofia: added sip_hangup_disposition variable + mod_sofia: allow send_message and notify events to send a message/notify without a body if needed. + mod_sofia: append -1 .. -N postfix after any X-headers as vars that have the same name + mod_sofia: cache auth_gateway_name in sofia for challenged bye + mod_sofia: cancel proxy or no-media mode if you purposely answer or pre_answer + mod_sofia: correct result code mapping for Unallocated Number (MODENDP-124) + mod_sofia: disable 100rel by default + mod_sofia: don't accept crypto in the RTP/AVP (MODENDP-126) + mod_sofia: don't put CN in sdp answer if it was not in the offer. + mod_sofia: fix Incorrect IP address shows up in SDP "o" field when multiple external IPs available and FS not bound to first (MODENDP-132) + mod_sofia: fix Wrong RTP media port destination after reinvite/UNHOLD (SFSIP-82) + mod_sofia: fix bug on linksys where they lie about the ptime and handle linksys transfer problem + mod_sofia: fix chat (send an IM) assumes that the user's profile is the same as their domain, which isn't necessarily so (SFSIP-83) + mod_sofia: fix dtmf handling of broken info dtmf endpoints + mod_sofia: fix eyebeam presence to be RFC compliant (MODENDP-144) + mod_sofia: fix ip change detection when in proxy mode + mod_sofia: fix register_proxy ignoring the paramaters (MODENDP-121) + mod_sofia: fix remote session refresh triggers request glare (MODENDP-131) + mod_sofia: fix rtp auto adjust running when it should not + mod_sofia: fix rtp sent to wrong port after some re-INVITE scenarios (MODENDP-141) + mod_sofia: fix sending of cn packets across bridge when we shouldn't + mod_sofia: fix sqlite issue with select of the sip contact + mod_sofia: fixed segfault on invalid presence payload + mod_sofia: gateway ping needs to look for 501 (SFSIP-78) + mod_sofia: handle multi contact register responses and register timeout better + mod_sofia: improve gateway resilience + mod_sofia: log ip and port you get reply to invite from + mod_sofia: make multiple-registations=true use the contact method and call-id option to do it the old way + mod_sofia: make proxy mode pull the port from m=image as well + mod_sofia: make register-proxy preserve the url composed from proxy but target the packets to desired address (MODENDP-121) + mod_sofia: many fixes for sonus rtp issues silence_when_idle=400 chanvar to send generated silence duing sleeps etc + mod_sofia: many fixes in presence handling + mod_sofia: passthrough t.38 fixes + mod_sofia: pick ipv4 or ipv6 based on sipip instead of having mixed in sdp + mod_sofia: send NOTIFY on TCP/UDP depending on the SUBSCRIBE (SFSIP-104) + mod_sofia: setting profile option multiple-registrations=contact key multi reg off the contact string + mod_sofia: wait for a reply on refer + mod_soundtouch: fixes and improvements, many options changed (MODAPP-149) + mod_soundtouch: updated to new module api + mod_spidermonkey: Segmentation fault in check_hangup_hook at mod_spidermonkey.c:1589 (MODLANG-74) + mod_spidermonkey: fix bug in apiExecute + mod_spidermonkey: fix memory pool handling and leaks + mod_spidermonkey: limit recursion busting through the stack (FSCORE-202) + mod_spidermonkey: make session.getVariable return blank string not the word false + mod_spidermonkey_curl: add optional content-type arg + mod_spidermonkey_odbc: fix numRows and add numCols + mod_spidermonkey_odbc: fix segfault (MODLANG-75) + mod_stress: new module for voice stress analysis + mod_syslog: don't log blank lines (FSCORE-163) + mod_tone_stream: let silence_stream://0 indicate perpetual silence + mod_vmd: add new module to detect voicemail "beep" + mod_voicemail: Add vm_alternate_greet_id param to directory entry (MODAPP-174) + mod_voicemail: Patch to add voicemail preference controlling date announcement new param 'play-date-announcement' to values 'first' 'last' or 'never' defaults to first to retain previous behavior (MODAPP-121) + mod_voicemail: Update mwi light after delete vm via web. (MODAPP-124) + mod_voicemail: add ability to get to options without listening to every saved message (MODAPP-115) + mod_voicemail: add ability to skip greeting when leaving a voicemail. (MODAPP-181) + mod_voicemail: add change-pass-key config file option + mod_voicemail: add forwarding support + mod_voicemail: add local dtmf driven alternat vm pass + mod_voicemail: add proper notification of a vm message being too short + mod_voicemail: add support for auth via a1-hash + mod_voicemail: add the "storage-dir" parameter to be set on a per-user basis (MODAPP-133) + mod_voicemail: add voicemail_greeting_path variable + mod_voicemail: added voicemail_alternate_greet_id variable + mod_voicemail: allow changing of password from voicemail to update user directory if using non-static config (MODAPP-156) + mod_voicemail: created email date (int overflow) (MODAPP-125) + mod_voicemail: don't try to deliver vm when no file was recorded. (MODAPP-133) + mod_voicemail: fix MWI with xml_curl used for directory (MODAPP-176) + mod_voicemail: fix Voicemail messages occasionally lost / stranded (MODAPP-178) + mod_voicemail: fix invalid event after message deleted (MODAPP-170) + mod_voicemail: fix mwi for phones with multiple registrations problem (MODAPP-153) + mod_voicemail: fix voicemail segfault on incorrect password (FSCORE-187) + mod_voicemail: fix voicemail_inject error handling (MODAPP-133) + mod_voicemail: fix voicemail_inject usage api call + mod_voicemail: improve error checking (MODAPP-142) + mod_voicemail: localize notification emails (MODAPP-139) + mod_voicemail: make more multi-domain friendly (MODAPP-162) + mod_voicemail: make playback created file macros optional (MODAPP-150) + mod_voicemail: recognize operator key in more places (MODAPP-159) + mod_voicemail: web interface displays incorrect created / last heard dates (MODAPP-123) + mod_wanpipe: removed + mod_xml_cdr: add https support + mod_xml_cdr: add optional a-leg prefix to xml cdr filenames (MDXMLINT-39) + mod_xml_cdr: add support for fallback webserver for cdr posting (FSCORE-238) + mod_xml_curl: Allow specification of HTTP method, and dynamic expansion of variables in URI. (MDXMLINT-41) + mod_xml_curl: added redirect following (max 10) + mod_xml_ldap: almost a complete rewrite of this module + mod_xml_rpc: allow setting of global realm without a global user (MDXMLINT-45) + mod_xml_rpc: fix multiple segfaults + mod_xml_rpc: fix segfault on originate via http + sofia-sip: updated to 1.12.10 (plus a few patches) + freeswitch (1.0.1) FIX: prevent intercept race condition that can also be solved with continue_on_fail=originator_cancel

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[commit] r10991 - in freeswitch/trunk/src: . include mod/endpoints/mod_sofia
by FreeSWITCH SVN 29 Dec '08

29 Dec '08

Author: anthm Date: Mon Dec 29 16:23:17 2008 New Revision: 10991 Log: add rtp_bug flags so we can keep track of all the bugs we have to emulate Modified: freeswitch/trunk/src/include/switch_rtp.h freeswitch/trunk/src/include/switch_types.h freeswitch/trunk/src/mod/endpoints/mod_sofia/mod_sofia.h freeswitch/trunk/src/mod/endpoints/mod_sofia/sofia_glue.c freeswitch/trunk/src/switch_rtp.c Modified: freeswitch/trunk/src/include/switch_rtp.h ============================================================================== --- freeswitch/trunk/src/include/switch_rtp.h (original) +++ freeswitch/trunk/src/include/switch_rtp.h Mon Dec 29 16:23:17 2008 @@ -431,6 +431,8 @@ SWITCH_DECLARE(switch_status_t) switch_rtp_activate_stun_ping(switch_rtp_t *rtp_session, const char *stun_ip, switch_port_t stun_port, uint32_t packet_count, switch_bool_t funny); +SWITCH_DECLARE(void) switch_rtp_intentional_bugs(switch_rtp_t *rtp_session, switch_rtp_bug_flag_t bugs); + /*! \} */ Modified: freeswitch/trunk/src/include/switch_types.h ============================================================================== --- freeswitch/trunk/src/include/switch_types.h (original) +++ freeswitch/trunk/src/include/switch_types.h Mon Dec 29 16:23:17 2008 @@ -460,6 +460,38 @@ } switch_rtp_flag_enum_t; typedef uint32_t switch_rtp_flag_t; +typedef enum { + RTP_BUG_NONE = 0, /* won't be using this one much ;) */ + + RTP_BUG_CISCO_SKIP_MARK_BIT_2833 = (1 << 0), + /* Some Cisco devices get mad when you send the mark bit on new 2833 because it makes + them flush their jitterbuffer and the dtmf along with it. + + This flag will disable the sending of the mark bit on the first DTMF packet. + */ + + + RTP_BUG_SONUS_SEND_INVALID_TIMESTAMP_2833 = (1 << 1) + /* + Sonus wrongly expects that, when sending a multi-packet 2833 DTMF event, The sender + should increment the RTP timestamp in each packet when, in reality, the sender should + send the same exact timestamp and increment the duration field in the 2833 payload. + This allows a reconstruction of the duration if any of the packets are lost. + + final_duration - initial_timestamp = total_samples + + However, if the duration value exceeds the space allocated (16 bits), The sender should increment + the timestamp one unit and reset the duration to 0. + + Always sending a duration of 0 with a new timestamp should be tolerated but is rarely intentional + and is mistakenly done by many devices. + The issue is that the Sonus expects everyone to do it this way instead of tolerating eiher way. + Sonus will actually ignore every packet with the same timestamp before concluding if it's DTMF. + + This flag will cause each packet to have a new timestamp. + */ + +} switch_rtp_bug_flag_t; #ifdef _MSC_VER #pragma pack(push, r1, 1) Modified: freeswitch/trunk/src/mod/endpoints/mod_sofia/mod_sofia.h ============================================================================== --- freeswitch/trunk/src/mod/endpoints/mod_sofia/mod_sofia.h (original) +++ freeswitch/trunk/src/mod/endpoints/mod_sofia/mod_sofia.h Mon Dec 29 16:23:17 2008 @@ -208,7 +208,7 @@ TFLAG_NOHUP = (1 << 18), TFLAG_NOSDP_REINVITE = (1 << 19), TFLAG_NAT = (1 << 20), - TFLAG_BUGGY_2833 = (1 << 21), + TFLAG_USEME = (1 << 21), TFLAG_SIP_HOLD = (1 << 22), TFLAG_INB_NOMEDIA = (1 << 23), TFLAG_LATE_NEGOTIATION = (1 << 24), @@ -552,6 +552,7 @@ uint32_t mismatch_count; uint32_t last_codec_ms; nua_event_t want_event; + switch_rtp_bug_flag_t rtp_bugs; }; struct callback_t { Modified: freeswitch/trunk/src/mod/endpoints/mod_sofia/sofia_glue.c ============================================================================== --- freeswitch/trunk/src/mod/endpoints/mod_sofia/sofia_glue.c (original) +++ freeswitch/trunk/src/mod/endpoints/mod_sofia/sofia_glue.c Mon Dec 29 16:23:17 2008 @@ -1775,10 +1775,6 @@ flags = (switch_rtp_flag_t) (SWITCH_RTP_FLAG_DATAWAIT); } - if (switch_test_flag(tech_pvt, TFLAG_BUGGY_2833)) { - flags |= SWITCH_RTP_FLAG_BUGGY_2833; - } - if ((val = switch_channel_get_variable(tech_pvt->channel, "dtmf_type"))) { if (!strcasecmp(val, "rfc2833")) { tech_pvt->dtmf_type = DTMF_2833; @@ -1923,7 +1919,9 @@ tech_pvt->ssrc = switch_rtp_get_ssrc(tech_pvt->rtp_session); switch_set_flag(tech_pvt, TFLAG_RTP); switch_set_flag(tech_pvt, TFLAG_IO); - + + switch_rtp_intentional_bugs(tech_pvt->rtp_session, tech_pvt->rtp_bugs); + if ((vad_in && inb) || (vad_out && !inb)) { switch_rtp_enable_vad(tech_pvt->rtp_session, tech_pvt->session, &tech_pvt->read_codec, SWITCH_VAD_FLAG_TALKING); switch_set_flag(tech_pvt, TFLAG_VAD); @@ -2205,12 +2203,12 @@ if ((tech_pvt->origin = switch_core_session_strdup(session, (char *) sdp->sdp_origin->o_username))) { if (strstr(tech_pvt->origin, "CiscoSystemsSIP-GW-UserAgent")) { - switch_set_flag_locked(tech_pvt, TFLAG_BUGGY_2833); + tech_pvt->rtp_bugs |= RTP_BUG_CISCO_SKIP_MARK_BIT_2833; switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "Activate Buggy RFC2833 Mode!\n"); } if (strstr(tech_pvt->origin, "Sonus_UAC")) { - switch_set_flag_locked(tech_pvt, TFLAG_BUGGY_2833); + tech_pvt->rtp_bugs |= RTP_BUG_SONUS_SEND_INVALID_TIMESTAMP_2833; switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_WARNING, "Hello,\nI see you have a Sonus!\n" "FYI, Sonus cannot follow the RFC on the proper way to send DTMF.\n" Modified: freeswitch/trunk/src/switch_rtp.c ============================================================================== --- freeswitch/trunk/src/switch_rtp.c (original) +++ freeswitch/trunk/src/switch_rtp.c Mon Dec 29 16:23:17 2008 @@ -199,6 +199,7 @@ switch_port_t stun_port; int from_auto; uint32_t cng_count; + switch_rtp_bug_flag_t rtp_bugs; }; static int global_init = 0; @@ -486,6 +487,11 @@ return port; } +SWITCH_DECLARE(void) switch_rtp_intentional_bugs(switch_rtp_t *rtp_session, switch_rtp_bug_flag_t bugs) +{ + rtp_session->rtp_bugs = bugs; +} + SWITCH_DECLARE(switch_status_t) switch_rtp_set_local_address(switch_rtp_t *rtp_session, const char *host, switch_port_t port, const char **err) { switch_socket_t *new_sock = NULL, *old_sock = NULL; @@ -1204,7 +1210,7 @@ rtp_session->dtmf_data.timestamp_dtmf, rtp_session->dtmf_data.out_digit_sofar, rtp_session->dtmf_data.out_digit_sub_sofar, rtp_session->dtmf_data.out_digit_dur, rtp_session->seq); - if (switch_test_flag(rtp_session, SWITCH_RTP_FLAG_BUGGY_2833)) { + if (rtp_session->rtp_bugs & RTP_BUG_SONUS_SEND_INVALID_TIMESTAMP_2833) { rtp_session->dtmf_data.timestamp_dtmf = rtp_session->last_write_ts + samples; } } @@ -1257,7 +1263,7 @@ switch_rtp_write_manual(rtp_session, rtp_session->dtmf_data.out_digit_packet, 4, - switch_test_flag(rtp_session, SWITCH_RTP_FLAG_BUGGY_2833) ? 0 : 1, + rtp_session->rtp_bugs & RTP_BUG_CISCO_SKIP_MARK_BIT_2833 ? 0 : 1, rtp_session->te, rtp_session->dtmf_data.timestamp_dtmf, &flags); switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "Send start packet for [%c] ts=%u dur=%d/%d/%d seq=%d\n",

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